With rapid development of Internet technologies, various new technologies and new services continuously appear, and a web (Web) technological change led by a hypertext markup language 5 (HTML5) technology gradually becomes a main service formation of the Internet. A web real-time communication (WebRTC) is a technology of performing real-time video and audio communication inside a browser, and the technology is based on a Web Hypertext Application Technology Working Group (WHATWG) protocol, to achieve a real-time communication (RTC) capability by providing a simple Java description language by using a browser. An ultimate objective of a WebRTC project is mainly to enable a Web developer to easily and quickly develop, based on a browser, abundant real-time multimedia applications, and the Web developer does not need to download or install any plug-in. The Web developer does not need to focus on a digital signal processing process of multimedia either, but only needs to write a simple Java language program. In addition, it is further intended that in the WebRTC project, a real-time communication platform can be established among multiple Internet browsers, to form a desirable ecosystem between a developer and a browser vendor.
Currently, each WebRtc application based on a mobile terminal includes a quality of service (QoS) improvement service, which improves QoS (for example: bandwidth) of a WebRtc application of user equipment. For example, when user equipment A and user equipment B perform WebRtc audio and video communication between each other, if it is found that user experience is relatively poor, the user equipment A needs to collect information about Internet Protocol (Internet Protocol, IP for short) addresses and service port numbers of the both parties in the communication, then send a service acceleration request to a QoS decision network element to improve QoS of an audio and video service of a current user, and send the collected information about the IP addresses and the service port numbers of the both parties in the communication to the QoS decision network element. The QoS decision network element forms a parameter by using the IP addresses and the service port numbers of the both parties in the communication, and sends an acceleration request to a policy and charging rule function entity (PCRF), to separately improve QoS of the user equipment A and QoS of the user equipment B, thereby improving user experience of audio and video communication performed between the user equipment A and the user equipment B.
The foregoing QoS improvement process of the WebRtc audio and video communications application in the prior art has the following problems: when user equipment A and user equipment B perform audio and video communication, if the user equipment A finds that user experience of the audio and video communication with the user equipment B is relatively poor, the user equipment A requests a QoS decision network element to perform QoS improvement. According to an existing QoS improvement processing method, the user equipment A can send a QoS improvement request to the QoS decision network element only after collecting information about IP addresses and service port numbers of the both parties that perform audio and video communication. The user equipment is responsible for collecting the IP addresses and the service port numbers of the both parties in the communication. Therefore, burden on the user equipment is relatively heavy, and difficulty in using the current QoS improvement method by a user is increased.